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@ -31,6 +31,9 @@
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#include "AmConferenceStatus.h"
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#include "AmConfig.h"
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#include "AmSessionContainer.h"
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#include "AmSessionScheduler.h"
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#include "sems.h"
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#include "log.h"
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@ -46,6 +49,7 @@ ConferenceFactory::ConferenceFactory(const string& _app_name)
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string ConferenceFactory::LonelyUserFile;
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string ConferenceFactory::JoinSound;
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string ConferenceFactory::DropSound;
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string ConferenceFactory::DialoutSuffix;
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int ConferenceFactory::onLoad()
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{
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@ -66,6 +70,12 @@ int ConferenceFactory::onLoad()
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JoinSound = cfg.getParameter("join_sound");
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DropSound = cfg.getParameter("drop_sound");
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DialoutSuffix = cfg.getParameter("dialout_suffix");
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if(DialoutSuffix.empty()){
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WARN("No dialout_suffix has been configured in the conference plug-in:\n");
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WARN("\t -> dial out will not be available\n");
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}
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return 0;
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}
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@ -74,13 +84,20 @@ AmSession* ConferenceFactory::onInvite(const AmSipRequest& req)
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return new ConferenceDialog(req.user);
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}
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ConferenceDialog::ConferenceDialog(const string& conf_id)
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: conf_id(conf_id), channel(0), play_list(this)
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ConferenceDialog::ConferenceDialog(const string& conf_id,
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AmConferenceChannel* dialout_channel)
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: conf_id(conf_id),
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channel(0),
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play_list(this),
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dialout_channel(dialout_channel),
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state(CS_normal)
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{
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dialedout = this->dialout_channel.get() != 0;
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}
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ConferenceDialog::~ConferenceDialog()
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{
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DBG("ConferenceDialog::~ConferenceDialog()\n");
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}
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void ConferenceDialog::onStart()
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@ -89,6 +106,16 @@ void ConferenceDialog::onStart()
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}
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void ConferenceDialog::onSessionStart(const AmSipRequest& req)
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{
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setupAudio();
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}
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void ConferenceDialog::onSessionStart(const AmSipReply& reply)
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{
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setupAudio();
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}
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void ConferenceDialog::setupAudio()
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{
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if(!ConferenceFactory::JoinSound.empty()) {
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@ -106,29 +133,39 @@ void ConferenceDialog::onSessionStart(const AmSipRequest& req)
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DropSound.reset(0);
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}
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channel.reset(AmConferenceStatus::getChannel(conf_id,getLocalTag()));
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play_list.close();// !!!
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if(dialout_channel.get()){
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play_list.addToPlaylist(new AmPlaylistItem(dialout_channel.get(),
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dialout_channel.get()));
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}
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else {
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channel.reset(AmConferenceStatus::getChannel(conf_id,getLocalTag()));
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play_list.addToPlaylist(new AmPlaylistItem(channel.get(),channel.get()));
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play_list.addToPlaylist(new AmPlaylistItem(channel.get(),
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channel.get()));
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}
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setInOut(&play_list,&play_list);
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setCallgroup(conf_id);
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//setInOut(channel.get(),channel.get());
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}
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void ConferenceDialog::onBye(const AmSipRequest& req)
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{
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play_list.close();
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setInOut(NULL,NULL);
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channel.reset(NULL);
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if(dialout_channel.get())
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disconnectDialout();
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closeChannels();
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setStopped();
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}
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void ConferenceDialog::process(AmEvent* ev)
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{
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ConferenceEvent* ce = dynamic_cast<ConferenceEvent*>(ev);
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if(ce){
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if(ce && (conf_id == ce->conf_id)){
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switch(ce->event_id){
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case ConfNewParticipant:
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@ -172,5 +209,257 @@ void ConferenceDialog::process(AmEvent* ev)
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return;
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}
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DialoutConfEvent* do_ev = dynamic_cast<DialoutConfEvent*>(ev);
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if(do_ev){
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if(dialedout){
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switch(do_ev->event_id){
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case DoConfConnect:
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connectMainChannel();
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break;
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case DoConfDisconnect:
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dlg.bye();
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closeChannels();
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setStopped();
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break;
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default:
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break;
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}
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}
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else {
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switch(do_ev->event_id){
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case DoConfDisconnect:
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connectMainChannel();
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break;
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case DoConfConnect:
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state = CS_dialout_connected;
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break;
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}
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}
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return;
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}
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AmSession::process(ev);
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}
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string dtmf2str(int event)
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{
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switch(event){
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case 0: case 1: case 2:
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case 3: case 4: case 5:
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case 6: case 7: case 8:
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case 9:
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return int2str(event);
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case 10: return "*";
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case 11: return "#";
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default: return "";
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}
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}
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void ConferenceDialog::onDtmf(int event, int duration)
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{
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DBG("ConferenceDialog::onDtmf\n");
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if(dialedout)
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return;
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switch(state){
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case CS_normal:
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dtmf_seq += dtmf2str(event);
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if(dtmf_seq.length() == 2){
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if(dtmf_seq == "#*")
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state = CS_dialing_out;
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dtmf_seq = "";
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}
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break;
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case CS_dialing_out:{
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string digit = dtmf2str(event);
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if(digit == "*"){
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if(!dtmf_seq.empty()){
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createDialoutParticipant("sip:" + dtmf_seq +
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ConferenceFactory::DialoutSuffix);
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state = CS_dialed_out;
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}
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else {
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state = CS_normal;
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}
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dtmf_seq = "";
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}
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else
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dtmf_seq += digit;
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} break;
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case CS_dialout_connected:
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if(event == 10){ // '*'
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AmSessionContainer::instance()
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->postEvent(dialout_id,
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new DialoutConfEvent(DoConfConnect,
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getLocalTag()));
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connectMainChannel();
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}
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break;
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case CS_dialed_out:
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if(event == 11){ // '#'
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disconnectDialout();
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}
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break;
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}
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}
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void ConferenceDialog::createDialoutParticipant(const string& uri)
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{
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dialout_channel.reset(AmConferenceStatus::getChannel(getLocalTag(),getLocalTag()));
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dialout_id = AmSession::getNewId();
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ConferenceDialog* dialout_session =
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new ConferenceDialog(conf_id,
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AmConferenceStatus::getChannel(getLocalTag(),
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dialout_id));
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AmSipDialog& dialout_dlg = dialout_session->dlg;
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dialout_dlg.local_tag = dialout_id;
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dialout_dlg.callid = AmSession::getNewId() + "@" + AmConfig::LocalIP;
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dialout_dlg.local_party = dlg.local_party;
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dialout_dlg.remote_party = uri;
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dialout_dlg.remote_uri = uri;
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string body;
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int local_port = dialout_session->rtp_str.getLocalPort();
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dialout_session->sdp.genRequest(AmConfig::LocalIP,local_port,body);
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dialout_dlg.sendRequest("INVITE","application/sdp",body,"");
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play_list.close(); // !!!
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play_list.addToPlaylist(new AmPlaylistItem(dialout_channel.get(),
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dialout_channel.get()));
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dialout_session->start();
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AmSessionContainer* sess_cont = AmSessionContainer::instance();
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sess_cont->addSession(dialout_id,dialout_session);
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}
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void ConferenceDialog::disconnectDialout()
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{
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if(dialedout){
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if(dialout_channel.get()){
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AmSessionContainer::instance()
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->postEvent(dialout_channel->getConfID(),
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new DialoutConfEvent(DoConfDisconnect,
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dialout_channel->getConfID()));
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}
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}
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else {
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AmSessionContainer::instance()
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->postEvent(dialout_id,
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new DialoutConfEvent(DoConfDisconnect,
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getLocalTag()));
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connectMainChannel();
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}
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}
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void ConferenceDialog::connectMainChannel()
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{
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dialout_id = "";
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dialout_channel.reset(NULL);
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play_list.close();
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if(!channel.get())
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channel.reset(AmConferenceStatus
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::getChannel(conf_id,
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getLocalTag()));
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play_list.addToPlaylist(new AmPlaylistItem(channel.get(),
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channel.get()));
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}
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void ConferenceDialog::closeChannels()
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{
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play_list.close();
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setInOut(NULL,NULL);
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channel.reset(NULL);
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dialout_channel.reset(NULL);
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}
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void ConferenceDialog::onSipReply(const AmSipReply& reply)
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{
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int status = dlg.getStatus();
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AmSession::onSipReply(reply);
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if(status < AmSipDialog::Connected){
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switch(dlg.getStatus()){
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case AmSipDialog::Connected:
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// connected!
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if(acceptAudio(reply.body,reply.hdrs)){
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ERROR("could not connect audio!!!\n");
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dlg.bye();
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setStopped();
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}
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else {
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if(getDetached() && !getStopped()){
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onSessionStart(reply);
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if(getInput() || getOutput())
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AmSessionScheduler::instance()->addSession(this,
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getCallgroup());
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else {
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ERROR("missing audio input and/or ouput.\n");
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}
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// send connect event
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AmSessionContainer::instance()
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->postEvent(dialout_channel->getConfID(),
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new DialoutConfEvent(DoConfConnect,
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dialout_channel->getConfID()));
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}
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}
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break;
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case AmSipDialog::Pending:
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switch(reply.code){
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case 180: break;//TODO: local ring tone.
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case 183: break;//TODO: remote ring tone.
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default: break;// continue waiting.
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}
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}
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}
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}
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